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Difference between 24bit/94k and FLAC?

post #1 of 12
Thread Starter 

What is the difference between them? 

post #2 of 12
FLAC is an encoding format, like MP3.

24bit is the bit depth of the recording. CD is 16bit for example.
96k (not 94 btw) is how many samples per second have been recorded.

FLAC can encode 24/96, and many others.

Hope that helps some.
post #3 of 12
Thread Starter 
Quote:
Originally Posted by m2man View Post

FLAC is an encoding format, like MP3.
24bit is the bit depth of the recording. CD is 16bit for example.
96k (not 94 btw) is how many samples per second have been recorded.
FLAC can encode 24/96, and many others.
Hope that helps some.
so 24bit/96k is better?
post #4 of 12
Quote:
Originally Posted by wolfetan44 View Post

so 24bit/96k is better?

 

There's no guarantee that just because the numbers are bigger -- greater bit-depth or higher sampling rate -- that the difference is audible to you or me or anyone else.

 

I'm not convinced at all, while others are. Differences of opinion on this subject are the cause of many flame wars across the Internet. HydrogenAudio is a good site for technical discussions.

post #5 of 12
Quote:
Originally Posted by wolfetan44 View Post

so 24bit/96k is better?

how can you "compare" them when they are not the same thing?

post #6 of 12
Thread Starter 
Quote:
Originally Posted by FrequencyBlue View Post

how can you "compare" them when they are not the same thing?

If one sounds better than the otherconfused.gif Also what is the best sounding audio files you can get?

post #7 of 12
Quote:
Originally Posted by wolfetan44 View Post

What is the difference between them? 

My two cents, hopefully I'm right.

 

Audio files are recorded (ripped from CDs) in Kilo bits per second (Kbit/s or kps?), like 128kbit/s, 192kbit/s, 256kbit/s, 320kbit/s, FLAC, etc.

Kilohertz (Khz) is not a recording format, it is a sample rate (measurement of signal bandwidth), examples like 16-bit/48Khz, 24-bit/96Khz, 24-bit/192Khz, 32-bit/384Khz, etc.


Edited by PurpleAngel - 8/27/12 at 2:11pm
post #8 of 12
Quote:
Originally Posted by PurpleAngel View Post

My two cents, hopefully I'm right.

 

Audio files are recorded in Kilo bits per second (Kbps or kps?), like 128kps, 192kps, 256kps, 320kps, FLAC, etc.

Kilohertz (Khz) is not a recording format, it is a sample rate (measurement of signal bandwidth), examples like 16-bit/48Khz, 24-bit/96Khz, 24-bit/192Khz, 32-bit/384Khz, etc.

 

Looks like this creates a lot of confusion, so let's try again:

 

1. All digital sound recordings use the same base concept: the analog sound wave is sampled at predetermined time intervals and amplitude values of each sample are recorded.

 

2. The sampling rate is the number of samples taken per second. The more samples, the better reflection of the original wave along the time axis. Considering that the audible frequencies extend to about 20kHz, to capture the shape of such wave you'd need several samples across it - this is quite important because sine wave sounds quite different from square or triangular.

 

3. The sample resolution or bit depth determines how many discrete states of the amplitude can be recorded in each sample. With 16-bit it's about 65 thousands. With 24-bit it's over 16 million.

 

4. Once we have all this data, it can be stored in a file in a number of ways, for example:

- just as captured: .WAV file does that

- compressed in a lossless way (so that the wave file can be reproduced exactly):  for example .FLAC

- with lossy compression (so it can be restored only approximately): .MP3 is such a format. Of course the more detailed the original (higher bit depth and sampling rate), the less damage will be caused by compression.

 

5. The kbps rate usually relates to streaming - i.e. data transfer rate between two devices in order to maintain continuity of processing. Kps is sometimes used to indicate the sampling rate - then it tells how many thousands of samples are taken per second. Nothe that sample is still 16 or 24-bit, so to convert it to kbps you need to multiply by bit depth and by number of channels.

post #9 of 12
Quote:
Originally Posted by wolfetan44 View Post

If one sounds better than the otherconfused.gif Also what is the best sounding audio files you can get?

 

 

Read this article, It will help to answer you question ...

 

http://people.xiph.org/~xiphmont/demo/neil-young.html


Edited by Lumos - 8/27/12 at 3:24am
post #10 of 12
Quote:
Originally Posted by wolfetan44 View Post

If one sounds better than the otherconfused.gif Also what is the best sounding audio files you can get?

 

Your standard CD is encoded with a sampling rate of 16 bit/44.1 khz. So if you are making flacs from CDs, there's no benefit in using a higher sampling rate. 

 

Flacs are lossless, so unless they are ripped poorly from a CD, both the original flac and CD will have the same audio quality. Same with using Apple Lossless format to rip a CD. 

 

Now it is possible to buy some downloadable flacs that have a higher resolution, but as others have pointed out, you may not be able to hear much/any difference. Try the B&W Society of Sound website.

post #11 of 12

In certain instances a recording can show more detail, air, better instrument separation/placement within the sound-stage on 24/96/192 files. It's not always the case but there are tracks that do exhibit a benefit. You also have to have gear that is up to par to show these differences.


Edited by lee730 - 8/27/12 at 7:46am
post #12 of 12
Quote:
Originally Posted by PleasantSounds View Post

2. The sampling rate is the number of samples taken per second. The more samples, the better reflection of the original wave along the time axis. Considering that the audible frequencies extend to about 20kHz, to capture the shape of such wave you'd need several samples across it - this is quite important because sine wave sounds quite different from square or triangular.

 

That's not how the sampling and reconstruction work, from theory or in practice.

 

To capture a 20 kHz sine wave, all you need is 40 kHz sampling rate—the sampling rate must at least double the frequency content of the original (see Shannon/Nyquist).  Go through all the steps, and what comes out when you play it back is a 20 KHz sine wave, just as it were.  To represent 20 kHz triangle or square wave perfectly, you need an infinite sampling rate or you lose some information.  The difference here is that a 20 kHz (fundamental) tone of something other than a pure sine wave contains frequencies above the fundamental: at 60 kHz, 100 kHz, and so on.  44.1 kHz sampling rate is not going to be able to capture that 60 kHz component or the others as they are; they would be aliased down.

 

So if you're interested in capturing ultrasonic frequencies, you need more samples.

 

There are practical issues associated with not having perfect brickwall filters and so on, but modern DACs are quite good at handling 44.1 kHz sampling rate without mucking up much of the frequencies below 22.05 kHz (and mostly just those at the very top, above 20 kHz).

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